Hey guys, Just in the middle of a Jabber roll out at the moment, and we are doing softphone only deployment with Windows 7. The problem we are having is that when you plug in the USB headset, Windows automatically makes it the default audio device, and all sounds are then sent through the headset only; You can't hear Jabber ringing unless you have your headset on. The fix to this is to open Windows sound preferences and set the internal sound card back as the "Default audio device" And select the headset as the "Default communications device" Everything then functions correctly.  Ringing through all devices. Phone calls through headset. I know this isn't really a jabber issue, but I thought many of you may have had the same issue previously.  Has anyone found a way around this? Is there a login script we can run, or use group policy potentially? Cheers! ... View more

Hi All, I have an H323 gateway that has multiple numbers delivered to it.  We are using overlap sending and the gateway is in Germany so we have variable length dial plans to deal with. I know this isnt a good design (I unfortuantely had no control over it) but it is how it is.  I now have been tasked with fixing it! We have an "important" number on 0 1234 567 0 We also have a number range of 0 1234 56 7000-7999 The problem we have is that users often cannot dial the users on the number range and the calls just get delivered to the "important" number. http://www.voiceie.com/cgi-bin/ultimatebb.cgi?ubb=print_topic;f=8;t=001144 I've found this around changing the interdigit timeout values on dial-peers / PSTN connections but  I'm not sure how it interacts with overlap sending etc, and changing it and testing isn't practical as we don't want to risk affecting users until we are sure. Our dial peers are configured with : destination-pattern 012345670 and destination-pattern 012345670.. If I add an interdigit timout to the E1 interface, I assume that PSTN inbound callers will have X Number of seconds to enter more digits after the last 0 - so if i set this to 5 or 6 that should be a reasonable amount of time to allow them to dial? ... View more

Guys, Am looking at an installation at the moment that I didn't do, I can't find anything wrong with it but it doesnt work properly. if you sign in using name@domain.com and allow the client to do proper service discovery, you only get phone services, no IM. If you manually enter the name of one of the IMP servers in the "Advanced" configuration, everything works fine. We are using a cisco-uds SRV record that comprises all of the call managers in our environment.  The UC Service profiles are all correctly configured with IMP, CTI, and voicemail servers. ... View more

Guys, I am working on a phone migration at the moment from an 8.6 cluster in mixed mode (Security mode 1) to a 11.0 cluster in non secure (security mode 0) 8.6 cluster: PHones have ITL and CTL 11.0 cluster: PHones have ITL only I followed the consolidation procedure from here: https://supportforums.cisco.com/document/60716/migrating-ip-phones-between-clusters-cucm-8-and-itl-files And phones can now move between clusters without issue. However, Ive noticed that they keep the CTL file with all the old servers in - even when registered to the new 11.0 CM. I  can see the ITLs update to the 11.0 cluster, but the CTL stays with the old names of the 8.6 cluster. Is this going to be an issue? I believe not, but does anyone have any experience of this? Thanks ... View more